WebRTC mandates that terminals in a WebRTC network support the Secure Real-time Transport Protocol (SRTP) and the Interactive Connectivity Establishment (ICE) framework in order to communicate. The mode of SRTP can be the Datagram Transport Layer Security (DTLS) protocol. If one of the participant devices does not support it the communication will fail.
For existing software IP (SIP) devices, whether SIP Devices in Enterprises or SIP Devices in Service Providers (IP Multimedia Subsystem (IMS) based or non-IMS based):
Most of them do not support ICE;
Most of them do not support SRTP. Although some support SRTP, most legacy ones support Session Description Protocol Security Descriptions (SDES), not the DTLS-SRTP; and
3GPP Standard (3GPP.33.328) chose SDES mode for SRTP.
If a WebRTC device wants to inter-connect with an IMS or a SIP network, it has to use a WebRTC-SIP Interworking Function, which consists of a Web Server, a SIP Interworking Function, and a Media Interworking Function, Besides the translation between HTTP and SIP signaling, this WebRTC-SIP Interworking Function needs to convert between SRTP and Real-time Transport Protocol (RTP) as well as ICE termination.
The introduction of WebRTC-SIP Interworking Function introduces another issue: When both ends of a communication are WebRTC enabled devices the media path still relays from the Media Interworking Function while WebRTC was designed for P2P purposes.
There is desired a solution to allow media to perform P2P communication when both ends are WebRTC enabled devices in an IMS or SIP network.